Digital Audio
Sample rate, also known as sampling frequency, is the number of times per second an analogue audio signal is measured during digital conversion. It determines the highest audio frequency that can be accurately captured according to the Nyquist theorem.
Sample rate defines how frequently an analogue waveform is measured when converting it into digital data using Pulse Code Modulation (PCM). The rate is expressed in hertz (Hz), meaning samples per second. Common professional values include 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, and 192 kHz.
According to the Nyquist theorem, a digital system must sample at least twice the highest frequency present in the signal to reproduce it faithfully. A 44.1 kHz sample rate captures audio up to around 22 kHz — slightly beyond human hearing — which is why it was adopted for Compact Disc audio.
Higher rates such as 96 kHz or 192 kHz are often chosen in professional production for greater precision during processing and more relaxed anti-aliasing filter design, though they demand higher storage and CPU resources. Conversely, lower rates such as 32 kHz or 22.05 kHz are used in broadcast or speech applications where bandwidth efficiency is more important than full-range fidelity.
Ultimately, the chosen sample rate represents a balance between audio quality, system performance, and compatibility. While very high rates are supported by modern hardware, 44.1 kHz remains the standard for music, and 48 kHz for video and broadcast production.
